Every inbound call I make from a PSTN phone to a SIP number living on my * box ends like this:
Scheduling destruction of call ‘email@example.com’ in 15000 ms
== Spawn extension (cp, 0, 1) exited non-zero on ‘SIP/000060002077777-3baa’
– Executing Hangup(“SIP/000060002077777-3baa”, “”) in new stack
== Spawn extension (cp, h, 1) exited non-zero on ‘SIP/000060002077777-3baa’
Before I added explicit hangups in my dialplan, I saw
== Spawn extension (cp, 000060002077777, 1) exited non-zero on ‘SIP/000060002077777-3243’
but I didn’t think I’d need to expicitly tell asterisk to hang up when one side or the other ends the call? As it stands, I have to do a “sip reload” and re-register before another incoming call will come through before asterisk re-registers itself automatically…I’m fairly new to asterisk, am I missing something? I’ve been perusing the docs and other people’s examples, but most people don’t seem to be setting explicit hangups…any advice would be gratefully received.